Method of and system for providing intelligent network control services in ip telephony

ABSTRACT

A method and system for providing intelligent network control services in IP telephony, wherein the system includes a location manager and an IP telephony proxy server. The location manager includes an interface to a legacy telephony service control entity and the IP telephony proxy server includes an IP interface to the location manager. If the IP telephony proxy server requires intelligent network services, then the IP telephony proxy server sends an IP telephony session initiation request to the called party at the location manager. The location manager uses the information to query the legacy telephony service control entity for routing information. When the location manager receives a routing response from the service control entity, the location manager maps the response to an IP telephony session control message back to the IP telephony proxy server.

BACKGROUND

The present invention relates generally to the field of IP telephony,and more particularly to providing intelligent network control servicesin IP telephony.

Internet telephony is the real-time delivery of voice, and othermultimedia data, between two or more parties across a network usingInternet protocols (IP). Internet telephony began in the mid-1990s withthe introduction of Internet phone software, Internet phone software isdesigned to run on a personal computer equipped with a sound card,speakers, microphone, and modem. Software compresses the voice signaland translates it into IP packets for transmission over the Internet.This basic PC-to-PC Internet telephony works, however, only if bothparties are using Internet phone software.

Internet telephony offers the opportunity to design a global multimediacommunications system that may eventually replace the existing circuitswitched telephony infrastructure. In a relatively short period of time,Internet telephony has advanced rapidly. Many software developers nowoffer PC telephony software.

Internet telephony is session based rather than connection based.Generally, a first Internet protocol, such as H.323 or SessionInitiation Protocol (SIP) is used to establish the session and negotiatethe capabilities for the session, and a second Internet protocol, suchReal-time Transport Protocol (RTP), is used to transport the actualmedia across the IP network.

While IP telephony offers benefits to both users and carriers in termsof cost and variety of media types, there is a substantial installedbase of traditional telephones served by the public switched telephonenetwork (PSTN). Moreover, in addition to its widespread nature, the PSTNoffers a rich set intelligent network services such as “800” numberservices, Virtual Private Network (ET) services, call forwarding, andthe like. Accordingly, IP telephony is not likely, anytime soon, toreplace the PSTN. However, there is a desire to integrate the PSTN withIP networks, including the Internet and private intranets. Thus, thereare instances when a call originated by a phone on the PSTN will berequired to be carried through an IP based network for eventual deliveryto a second phone on the PSTN. There is a further desire to provide allof the intelligent network services that currently exist in the PSTN toIP telephony calls.

SUMMARY

The present invention provides a method of and system for providingintelligent network control services in IP telephony. The systemincludes a location manager and an IP telephony proxy server. Thelocation manager includes an interface to a legacy telephony servicecontrol entity, such as service control point (SCP). The IP telephonyproxy server, which may be, for example, an H.323 gatekeeper or aSession Initiation Protocol (SIP) proxy server, includes an IP interfaceto the location manager.

When the IP telephony proxy server receives a request to initiate an IPtelephony session or call to a called party address or number, the IPtelephony proxy server determines if it needs intelligent networkservices in order to route the request to the called party address ornumber. Examples of sessions requiring intelligent network services are“800” calls and virtual private network (VNET) calls. If the IPtelephony proxy server requires intelligent network services, the IPtelephony proxy server sends an IP telephony session initiation requestto the called party at the location manager. The location manager usesthe information received from the IP telephony proxy server to query thelegacy telephony service control entity for routing information. Whenthe location manager receives a routing response from the servicecontrol entity, the location manager maps the response to an IPtelephony session control message back to the IE telephony proxy server.

BRIEF DESCRIPTION OF THE DRAWINGS

FIG. 1 is a block diagram of a system according to the presentinvention.

FIG. 2 is a call flow diagram of processing of a virtual private network(VNET) call according to one embodiment of the present invention.

FIG. 3 is a call flow diagram of processing of a virtual private network(VNET) call according to an alternative embodiment of the presentinvention.

DETAILED DESCRIPTION

Referring now to the drawings, and first to FIG. 1, a system accordingto present invention is designated generally by the numeral 11. Thesystem 11 is adapted to provide telephony services between and amongsubscribers using traditional telephones 13 and Internet telephones 15.The signaling and media for calls according to the present invention aretransported at least in part over the Internet, indicated generally at17.

Traditional telephones 13 are connected to Internet 17 throughtraditional telephone switching equipment, such as PBXs 19 and IPtelephony gateways 21. IP telephony gateways 21 each include a signalinggateway (not shown) and a media gateway (not shown). The signalinggateway provides bidirectional translation between PSTN telephonysignaling, such as SS7, messages and IP telephony signaling messages inprotocols such as H.323 or Session Initiation Protocol (SIP). Typically,media gateways use one IP signaling protocol or the other, but not both.The media gateway provides bidirectional conversion between timedivision multiplexed (TDM) signals and IP transport packets in aprotocol such as real-time transport protocol (RTP). IP phones 15 may beconnected directly to be Internet through a local area network or bymodem connection through an Internet service provider.

Generally, call signaling and media are transported across Internet 17between an ingress IP telephony gateway 21 a and an egress IP telephonygateway 21 b. Typically, routing information is supplied by a proxyserver, such as a SIP proxy server 23 or an H.323 gatekeeper 25. In theSIP protocol, and invite message is sent from the ingress IP telephonygateway 21 a to the address of the called party at the SIP proxy server23. For normal calls that do not require intelligent network services,the SIP proxy server 23 knows the address of the called party at theegress IP telephony gateway 21 b. IP call setup signaling messages aretransported back and forth between the IP telephony gateways 21 and theSIP proxy server 23 until the call is setup. In the present invention,the SIP proxy server 23 and the H.323 gatekeeper 25 are combined in acall control entity 27.

The proxy servers 23 and 25 cannot, by themselves, handle calls thatrequire intelligent network services. Examples of calls requiringintelligent network services are “800” number calls, Virtual PrivateNetwork (VNET) calls, local number portable calls, call forwarded calls,and the like. In traditional PSTN telephony, switches consult servicecontrol entities, known as service control points (SCPs), for routinginformation, such as number translation, in order to route callsproperly.

The system 11 makes use of a legacy PSTN system service control entityindicated generally at 29, which may be an SCP. The system 11 includes alocation manager 31, which provides alias resolution, gateway selection,and mobility management services to the call control entity 27, as wellas accessing the service control entity 29 for such services as VNET andlocal number portability services on behalf of the call control entity27.

The location manager 31 functions as a SIP redirect server. A redirectserver is a server that accepts a SIP request, maps the address intozero or more new addresses and returns are these addresses to theclient. Unlike a SIP proxy server, a redirect server does not initiateits own SIP requests. Unlike a SIP user agent server, a redirect serverdoes not accept calls. Thus, if a server of the call control entity 27cannot send a session initiation request to the IP telephony gateway 21,then the server sends a session initiation request to the called partyat the location manager 31. The location manager 31 either consults itsown database or accesses the legacy service control entity 29 to obtaina new address for the called party. The location manager 31 then returnsthe new address to the appropriate server of the call control entity 27.

In a preferred embodiment of the present invention, the H.323 gatekeeper25 is modified to function in the SIP protocol. Thus, the H.323gatekeeper 25 communicates with H.323 IP telephony gateways and Internetappliances in the H.323 protocol, and with SIP IP telephony gateways,Internet appliances, and the location manager 31 in the SIP protocol.

Referring now to FIG. 2, the PBX 19 a sends a setup message 33 to the IPtelephony gateway 21 a. The IP telephony gateway 21 a maps of the setupmessage 33 into a SIP invite request 35 addressed to the SIP proxyserver 23. The SIP proxy server 23 is unable by itself to process setupfor a VNET call. Accordingly, the SIP proxy server 23 sends a SIP inviterequest 37 to the dialed number at the location manager 31.

Upon receipt of the invite request 37, the location manager 31 queriesthe service control entity 29 with a routing request 39. The servicecontrol entity 29 performs a data lookup and responds to the locationmanager 31 with a routing response 41. The location manager 31 mapsresponse 41 into a SIP temporarily moved response 43 directed back toSIP proxy server 23. As is well-known those skilled in the art, SIPresponses are identified by a number, which for the case of thetemporarily moved response is 302. The response 43 provides the SIPproxy server 23 with an IP address for the called party at the egress IPtelephony gateway 21 b. Accordingly, the SIP proxy server 23 sends aninvite request 45 to the called party at the egress IP telephony gateway21 b.

Upon receipt of the invite 45, the egress IP telephony gateway 21 bsends a setup message 47 to the PBX 19 b. When the PBX 19 b rings theline of the called party, the PBX 19 b sends an alerting message 49 backto the egress IP telephony gateway 21 b. The egress IP telephony gateway21 b then sends a SIP 180 ringing message 51 back to the SIP proxyserver 23, which in turn sends a SIP 180 ringing response 53 to theingress IP telephony gateway 21 a. The ingress IP telephony gateway 21 athen sends an alerting message 55 to the PBX 19 a, which provides aringing tone to the caller party. When the called party answers, PBX 19b sends a connect message 57 to the egress IP telephony gateway 21 b.The egress IP telephony gateway 21 b in turn sends a SIP 200 OK response59 to the SIP proxy server 23. The proxy server 23 sends a 200 OKresponse 61 to the ingress IP telephony gateway 21 a. Upon receipt ofthe response 61, the ingress IP telephony gateway 21 a sends a connectmessage 63 to the PBX 19 a and a SIP ACK request 65 to the SIP proxyserver 23. The SIP proxy server 23 sends an ACK request 67 to the egressIP telephony gateway 21 b and the VNET session is established.

At the conclusion of the VNET session, the called party hangs up and thePBX 19 b sends a release message 69 to the egress IP telephony gateway21 b. The egress IP telephony gateway 21 b maps release 69 into a SIPBYE request 71 addressed to the calling party at SIP proxy server 23.The SIP proxy server 23 then sends a BYE request 73 to the calling partyat the ingress IP telephony gateway 21 a. The ingress if telephonygateway 21 a sends a release message 75 to the PBX 19 a to terminate thecall. The ingress IP telephony gateway 21 a also sends an ACK request 77to the SIP proxy server 23. The SIP proxy server 23 sends an ACK request79 back to the egress IP gateway 21 b. The SIP proxy server 23 alsosends a session detail record 81 to an appropriate billing authority.

Referring now to FIG. 3, in which the signaling gateway of the ingressIP telephony gateway 21 a uses the H.323 protocol. The PBX 19 a sends asetup message 83 to the ingress IP telephony gateway 21 a. The ingressIP telephony gateway 21 a maps the setup message 83 into an H.323 ARQmessage 85 addressed to the H.323 gatekeeper 25. The H.323 gatekeeper 25responds to message 85 with an H.323 ACF message 87. Upon receipt ofmessage 87, the ingress if telephony gateway 21 a sends an H.323 setupmessage 89 to gatekeeper 25. The H.323 gatekeeper 25 is unable by itselfto process setup for a VNET call. Accordingly, the H.323 gatekeeper 25sends a SIP invite request 91 to the dialed number at the locationmanager 31.

Upon receipt of invite request 91, the location manager 31 queries theservice control entity 29 with routing request 93. The service controlentity 29 performs a data lookup and responds to the location manager 31with a routing response 95. The location manager 31 determines that thecall should be routed to the called party at the egress IP telephonygateway 21 b and sends a SIP 305 temporarily moved response 97 back tothe H.323 gatekeeper 25. The H.323 gatekeeper 25 sends a SIP inviterequest 99 to the called party at the egress IP telephony gateway 21 b.Upon receipt of the SIP invite request 99, the egress IP telephonygateway 21 b sends a setup message 101 to the PBX 19 b. When the PBX 19b rings the line of the called party, the PBX 19 b sends an alertingmessage 103 back to the egress IP telephony gateway 21 b. The egress IPtelephony gateway 21 b then sends a SIP 180 ringing message 105 back tothe H.323 gatekeeper 25, which in turn sends a H.323 alerting message107 to the ingress IP telephony gateway 21 a. The ingress IP telephonygateway 21 a then sends an alerting message 109 to PBX 19 a, whichprovides a ringing tone to the calling party. When the called partyanswers, the PBX 19 b sends a connect message 11 to the egress IPtelephony gateway 21 b. The egress IP telephony gateway 21 b in turnsends a SIP 200 OK response 113 to the H.323 gatekeeper 25. The H.323gatekeeper 25 sends an H.323 connect message 115 to the ingress IPtelephony gateway 21 a and a SIP ACK request 116 back to the egress IPtelephony gateway 21 b. Upon receipt of the message 115, the ingress IPtelephony gateway 21 a sends a connect message 117 to the PBX 19 a andthe VNET session is established.

At the conclusion of the VNET session, the called party hangs up and thePBX 19 b sends a release message 119 to gateway 21 b. Gateway 21 b mapsrelease 119 into a SIP BYE request 121 addressed to the calling party atthe H.323 gatekeeper 25. The H.323 gatekeeper 25 then sends an H.323release message 123 to the calling party at the ingress IP telephonygateway 21 a and a SIP ACK request 124 back to the egress IP telephonygateway 21 b. The ingress IP telephony gateway 21 a sends a releasemessage 125 to the PBX 19 a to terminate the call. According to theH.323 protocol, at the conclusion of the session, the H.323 gatekeeper25 sends a disengage request 127 to the ingress IP telephony gateway 21a, which responds with a disengage confirm 129. Then the H.323gatekeeper 25 sends an end session command 131 to the ingress IPtelephony gateway 21 a, which responds with an end session command ACK133. The H.323 gatekeeper 25 then sends a session detail record 135 tothe appropriate billing authority.

From the foregoing, it may be seen that the present invention provides amethod and system for providing intelligent network services in an IPtelephony system. The location manager of the present inventionfunctions as a SIP redirect server to provide signaling routinginformation to proxy servers. Those skilled in the art will recognizealternative embodiments given the benefit of the foregoing disclosure.Accordingly, the foregoing is intended for purposes of illustration andnot of limitation.

1. (canceled)
 2. A system comprising: a location manager to: receive afirst Session Initiation Protocol (SIP) invite request from a SIP proxydevice, acquire a first address for a first called party in response toreceiving the first SIP invite request, send the first address to theSIP proxy device, receive a second SIP invite request from an H.323device, acquire a second address for a second called party in responseto receiving the second SIP invite request, and send the second addressto the H.323 device.
 3. The system of claim 2, where the first SIPinvite request is associated with an “800” call, a virtual privatenetwork (VNET) call, a local number portability request, or a callforwarding request.
 4. The system of claim 2, where the second SIPinvite request is associated with an “800” call, a virtual privatenetwork (VNET) call, a local number portability request, or a callforwarding request.
 5. The system of claim 2, where the location managerfunctions as a SIP redirect server.
 6. The system of claim 2, where,when acquiring a first address for a first called party, the locationmanager is configured to: acquire the first address from a localdatabase.
 7. The system of claim 2, where, when acquiring a firstaddress for a first called party, the location manager is configured to:acquire the first address from a legacy Public Switched TelephoneNetwork system service control entity.
 8. The system of claim 2, wherethe first address includes an address of the first called party at afirst egress Internet Protocol (IP) gateway, and where the secondaddress includes an address of the second called party at a secondegress IP gateway.
 9. A method comprising: receiving a first SessionInitiation Protocol (SIP) invite request from a SIP proxy device;obtaining a first address for a first called party in response toreceiving the first SIP invite request; sending the first address to theSIP proxy device in a first SIP response message; receiving a second SIPinvite request from an H.323 device; obtaining a second address for asecond called party in response to receiving the second SIP inviterequest; and sending the second address to the H.323 device in a secondSIP response message.
 10. The method of claim 9, where at least one ofthe first SIP invite request or the second SIP invite request relates toan “800” call.
 11. The method of claim 9, where at least one of thefirst SIP invite request or the second SIP invite request relates to avirtual private network (VNET) call.
 12. The method of claim 9, where atleast one of the first SIP invite request or the second SIP inviterequest relates to a local number portability request.
 13. The method ofclaim 9, where at least one of the first SIP invite request or thesecond SIP invite request relates to a call forwarding request.
 14. Themethod of claim 9, where at least one of the obtaining a first addressor the obtaining a second address includes: sending a routing request toa legacy Public Switched Telephone Network system service controlentity.
 15. The method of claim 9, where at least one of the obtaining afirst address or the obtaining a second address includes: accessing alocal database.
 16. The method of claim 9, where at least one of thefirst SIP response message or the second SIP response message includes aSIP 302 “temporarily moved” response message.
 17. A method comprising:obtaining, in response to receiving a first Session Initiation Protocol(SIP) invite request from a SIP proxy device, a first address for afirst called party at a first egress Internet Protocol (IP) gateway;sending the first address to the SIP proxy device in a first SIPresponse message; obtaining, in response to receiving a second SIPinvite request from an H.323 device, a second address for a secondcalled party at a second egress IP gateway; and sending the secondaddress to the H.323 device in a second SIP response message.
 18. Themethod of claim 17, where at least one of the first SIP invite requestor the second SIP invite request relates to an “800” call or a virtualprivate network (VNET) call.
 19. The method of claim 17, where at leastone of the first SIP invite request or the second SIP invite requestrelates to a local number portability request or a call forwardingrequest.
 20. The method of claim 17, where at least one of the obtaininga first address or the obtaining a second address includes: sending arouting request to a legacy Public Switched Telephone Network systemservice control entity.
 21. The method of claim 17, where at least oneof the obtaining a first address or the obtaining a second addressincludes: accessing a local database.